Split webrtc/video into webrtc/{audio,call,video}.

webrtc/webrtc.gyp

 # Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
 #
 # Use of this source code is governed by a BSD-style license
 # that can be found in the LICENSE file in the root of the source
 # tree. An additional intellectual property rights grant can be found
 # in the file PATENTS.  All contributing project authors may
 # be found in the AUTHORS file in the root of the source tree.
 {
   'conditions': [
     ['include_tests==1', {
       'includes': [
         'libjingle/xmllite/xmllite_tests.gypi',
         'libjingle/xmpp/xmpp_tests.gypi',
         'p2p/p2p_tests.gypi',
         'sound/sound_tests.gypi',
         'webrtc_tests.gypi',
       ],
     }],
     ['enable_protobuf==1', {
       'targets': [
         {
           # This target should only be built if enable_protobuf is defined
           'target_name': 'rtc_event_log_proto',
           'type': 'static_library',
-          'sources': ['video/rtc_event_log.proto',],
+          'sources': ['call/rtc_event_log.proto',],
           'variables': {
-            'proto_in_dir': 'video',
-            'proto_out_dir': 'webrtc/video',
+            'proto_in_dir': 'call',
+            'proto_out_dir': 'webrtc/call',
           },
         'includes': ['build/protoc.gypi'],
         },
       ],
     }],
     ['include_tests==1 and enable_protobuf==1', {
       'targets': [
         {
           'target_name': 'rtc_event_log2rtp_dump',
           'type': 'executable',
-          'sources': ['video/rtc_event_log2rtp_dump.cc',],
+          'sources': ['call/rtc_event_log2rtp_dump.cc',],
           'dependencies': [
             '<(DEPTH)/third_party/gflags/gflags.gyp:gflags',
             'rtc_event_log',
             'rtc_event_log_proto',
             'test/test.gyp:rtp_test_utils'
           ],
         },
       ],
     }],
   ],
   'includes': [
     'build/common.gypi',
+    'audio/webrtc_audio.gypi',
+    'call/webrtc_call.gypi',
     'video/webrtc_video.gypi',
   ],
   'variables': {
     'webrtc_all_dependencies': [
       'base/base.gyp:*',
       'sound/sound.gyp:*',
       'common.gyp:*',
       'common_audio/common_audio.gyp:*',
       'common_video/common_video.gyp:*',
       'modules/modules.gyp:*',
@@ skipping 36 matching lines @@
         'audio_send_stream.h',
         'call.h',
         'config.h',
         'frame_callback.h',
         'stream.h',
         'transport.h',
         'video_receive_stream.h',
         'video_renderer.h',
         'video_send_stream.h',
 
+        '<@(webrtc_audio_sources)',
+        '<@(webrtc_call_sources)',
         '<@(webrtc_video_sources)',
       ],
       'dependencies': [
         'common.gyp:*',
+        '<@(webrtc_audio_dependencies)',
+        '<@(webrtc_call_dependencies)',
         '<@(webrtc_video_dependencies)',
         'rtc_event_log',
       ],
       'conditions': [
         # TODO(andresp): Chromium libpeerconnection should link directly with
         # this and no if conditions should be needed on webrtc build files.
         ['build_with_chromium==1', {
           'dependencies': [
             '<(webrtc_root)/modules/modules.gyp:video_capture',
             '<(webrtc_root)/modules/modules.gyp:video_render',
           ],
         }],
       ],
     },
     {
       'target_name': 'rtc_event_log',
       'type': 'static_library',
       'sources': [
-        'video/rtc_event_log.cc',
-        'video/rtc_event_log.h',
+        'call/rtc_event_log.cc',
+        'call/rtc_event_log.h',
       ],
       'conditions': [
         # If enable_protobuf is defined, we want to compile the protobuf
         # and add rtc_event_log.pb.h and rtc_event_log.pb.cc to the sources.
         ['enable_protobuf==1', {
           'dependencies': [
             'rtc_event_log_proto',
           ],
           'defines': [
             'ENABLE_RTC_EVENT_LOG',
           ],
         }],
       ],
     },
 
   ],
 }